|  | /* | 
|  | *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved. | 
|  | * | 
|  | *  Use of this source code is governed by a BSD-style license | 
|  | *  that can be found in the LICENSE file in the root of the source | 
|  | *  tree. An additional intellectual property rights grant can be found | 
|  | *  in the file PATENTS.  All contributing project authors may | 
|  | *  be found in the AUTHORS file in the root of the source tree. | 
|  | */ | 
|  | #ifndef CALL_CALL_CONFIG_H_ | 
|  | #define CALL_CALL_CONFIG_H_ | 
|  |  | 
|  | #include <memory> | 
|  | #include <optional> | 
|  |  | 
|  | #include "api/environment/environment.h" | 
|  | #include "api/fec_controller.h" | 
|  | #include "api/metronome/metronome.h" | 
|  | #include "api/neteq/neteq_factory.h" | 
|  | #include "api/network_state_predictor.h" | 
|  | #include "api/scoped_refptr.h" | 
|  | #include "api/task_queue/task_queue_base.h" | 
|  | #include "api/transport/bitrate_settings.h" | 
|  | #include "api/transport/network_control.h" | 
|  | #include "api/units/time_delta.h" | 
|  | #include "call/audio_state.h" | 
|  | #include "call/rtp_transport_config.h" | 
|  | #include "call/rtp_transport_controller_send_factory_interface.h" | 
|  |  | 
|  | namespace webrtc { | 
|  |  | 
|  | class AudioProcessing; | 
|  |  | 
|  | struct CallConfig { | 
|  | // If `network_task_queue` is set to nullptr, Call will assume that network | 
|  | // related callbacks will be made on the same TQ as the Call instance was | 
|  | // constructed on. | 
|  | explicit CallConfig(const Environment& env, | 
|  | TaskQueueBase* network_task_queue = nullptr); | 
|  |  | 
|  | // Move-only. | 
|  | CallConfig(CallConfig&&) = default; | 
|  | CallConfig& operator=(CallConfig&& other) = default; | 
|  |  | 
|  | ~CallConfig(); | 
|  |  | 
|  | RtpTransportConfig ExtractTransportConfig() const; | 
|  |  | 
|  | Environment env; | 
|  |  | 
|  | // Bitrate config used until valid bitrate estimates are calculated. Also | 
|  | // used to cap total bitrate used. This comes from the remote connection. | 
|  | BitrateConstraints bitrate_config; | 
|  |  | 
|  | // AudioState which is possibly shared between multiple calls. | 
|  | scoped_refptr<AudioState> audio_state; | 
|  |  | 
|  | // Audio Processing Module to be used in this call. | 
|  | AudioProcessing* audio_processing = nullptr; | 
|  |  | 
|  | // FecController to use for this call. | 
|  | FecControllerFactoryInterface* fec_controller_factory = nullptr; | 
|  |  | 
|  | // NetworkStatePredictor to use for this call. | 
|  | NetworkStatePredictorFactoryInterface* network_state_predictor_factory = | 
|  | nullptr; | 
|  |  | 
|  | // Call-specific Network controller factory to use. If this is set, it | 
|  | // takes precedence over network_controller_factory. | 
|  | std::unique_ptr<NetworkControllerFactoryInterface> | 
|  | per_call_network_controller_factory; | 
|  | // Network controller factory to use for this call if | 
|  | // per_call_network_controller_factory is null. | 
|  | NetworkControllerFactoryInterface* network_controller_factory = nullptr; | 
|  |  | 
|  | // NetEq factory to use for this call. | 
|  | NetEqFactory* neteq_factory = nullptr; | 
|  |  | 
|  | TaskQueueBase* network_task_queue_ = nullptr; | 
|  | // RtpTransportControllerSend to use for this call. | 
|  | RtpTransportControllerSendFactoryInterface* | 
|  | rtp_transport_controller_send_factory = nullptr; | 
|  |  | 
|  | Metronome* decode_metronome = nullptr; | 
|  | Metronome* encode_metronome = nullptr; | 
|  |  | 
|  | // The burst interval of the pacer, see TaskQueuePacedSender constructor. | 
|  | std::optional<TimeDelta> pacer_burst_interval; | 
|  |  | 
|  | // Enables send packet batching from the egress RTP sender. | 
|  | bool enable_send_packet_batching = false; | 
|  | }; | 
|  |  | 
|  | }  // namespace webrtc | 
|  |  | 
|  | #endif  // CALL_CALL_CONFIG_H_ |